During the transmission of digitized signals, the latter are numerically coded in the transmitter, then decoded in a receiver for their reproduction. The present invention deals with the antinomy between on the one hand, the search for a transmission quality that generally brings about, for a set rate of thruput, a relatively long coding and decoding delay and, on the other hand, the coding and decoding delay that, in some applications must be short.
In the present description, there is called coding/decoding delay the time length that separates the input of a sample into the coding device from the output of the corresponding sample at the decoding device. In order to be free from the particular execution of the coding process and/or from the structure of the circuits permitting this coding, it will be considered that the computations done at the time of these processes are infinitely fast in the coding as well as in the decoding machine. There are thus involved, in the computations of the coding/decoding time lag, only parameters such as the length of time for of acquiring numerical signal rasters, the delay imposed by a filter bank, and/or the time corresponding to a multiplexing of the samples.
In the case of a transform-type coding device, this delay will exceed the duration of a coded raster added to the delay developed by the transform. In the case of a low-delay coding device of the LD-CELP type, such as that described by J. H. Chen et al in the article titled "A low delay CELP coder for CCITT 16 kb/s speed coding standard", published in IEEE J. Sel. Areas Commun. Vol. 10, pp 830-849, the delay is linked to the five samples that constitute a basic raster. It will be noted that a coding diagram has a delay expressed in number of samples. In order to extract from this a time value, there must be brought into play the sampling frequency at which the coder is used, according to the relation: EQU time duration=delay in samples/sampling frequency
As for the coding quality, this is a parameter difficult to define, knowing that the final receiver, that is to say the hearer's ear, cannot give precise quantitative results. Furthermore, measurements such as that of the signal to noise ratio, are not relevant because they do not take into account the psycho-acoustical masking properties of the auditory system. Statistical techniques such as those recommended by the notice ITU-R-BS-1116, permit to separate different coding algorithms with respect to coding quality.
It will be noted, however, that an improvement of the signal to noise ratio achieved on the frequency aggregate of the sound signal, makes it possible to ensure an improvement of the perceived quality.
The coding systems of generic audio-numerical signals, that is to say without hypothesis regarding the mode of production of these signals, until now, have not seriously considered as a constraint the matter of the signal reconstruction delay. One exception however is illustrated by the process described by F. Rumseyi in the article titled "Hearing both sides-stereo sound for TV in the UK" published in IEE review, vol. 36, No. 5, pp 173-176. In this process, however, the compression levels reached do not permit to compete with the coders with classical transforms.
Among the algorithms that are standardized by ISO (ISO/IEC 13818-3) the minimal reconstruction delays range from 18 ms for the simplest coder--and therefore the least efficient one--to more than 100 ms for the most complex coder. Other coding processes not standardized by ISO, such as the so-called ASPEC (Adaptative Spectral Perceptual Entropy coding) process described by K. Brandenburg et al, or the so-called ATRAC process (Adaptative Transform Acoustic Coding) described by K. Tsutsui typically present coding/decoding delays of the order of approximately one hundred milliseconds.
The efficiency of the coding system is bound to the side of the filterbanks that are generally used, to the taking into account the long term redundancies in the signals to be coded, to the optimal distribution of the binary allocations over a duration longer than the raster, etc. Taking into account these elements at coding time has as a consequence to increase the delay of the coding/decoding system.
It will be noted that the low delay coders often are related to the speech coding for telephone duplex connections, for example, or to be associated with echo cancelers. Designed most often for sample frequencies of 8 kHz to 16 kHz, their quality level proves insufficient to code generic audio-numerical signals in a manner close to the original.
The purpose of the present invention is to propose, within this context, a coding system and the associated decoding system, that permits the receiving side simultaneously to reconstruct a quality audio-numerical signal, and a lesser quality audio-numerical signal with a coding/decoding delay of which is as low as possible.
Such a coding/decoding system is already known and there must be mentioned the Preprint 4132 of the 99th AES Convention of October 1995 in New York, at which Bernhard Grill et al describe hierarchical audio-numerical coding systems, that is to say systems the output bit flow of which comprises a sub-group of bits that may permit a decoding and reconstitution of a significant or pertinent sound signal, but with a low quality compared to that obtained by decoding and reconstitution of the total bit flow.
Such coding systems comprise a coder to code a high quality sound signal the output of which is connected to the input of a decoder, and a difference circuit that performs the difference between the signal obtained at the output of the decoder and the original signal. The difference signal itself is subject, in a second stage, to similar coding, decoding, and difference computation treatments. The third stage codes the difference residual signal. The signals coming out of the coders of the three stages then are multiplexed so as to form a hierarchical numerical flow. Several modes of execution are presented, one of which specifies that, in the first stage, the coder is a low bit output coder with a relatively low coding delay. The coder of the second stage, however, is a longer delay coder.
With such a system there are thus available three flows multiplexed into a single output flow, one of these flows being developed with the low delay coder presenting a low delay and a lower quality level, while the other two show higher delays but bring in the flow of information necessary to a high quality reproduction.
In the systems presented by Bernhard Grill, however, each coder is, in reality, constituted by a under-sampled filterbank and a coder. Likewise, each decoder in reality is made up of a decoder, of a filterbank associated with the filterbank of the coder and that is over-sampling. It has been possible to observe that the use of such coders and decoders in this particular structure still brings about a relatively high coding/decoding delay of the low quality flow.